Grandstream UCM6200
From VoIP.ms Wiki
Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video conferencing, video surveillance, data tools, mobility options, and facility access management onto one common network that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.
These boxes typically support a phone system with up to 500 endpoints carrying up to fifty simultaneous calls. There are also a limited number of analogue telephone adapter ports (ranging from two for the UCM6202 to eight for the UCM6208) with provision to use analogue landlines as a fallback. The system is suited for a small to mid-size hotel or commercial office, offering features such as call transfer, conference, forwarding, call recording and toll restriction; there's even an automated wake-up call for hotel clients.
This page covers only basic information to establish a connection between the Grandstream PBX and one or more of the VoIP.ms servers. It is also necessary to perform many other configuration tasks, such as creating connections from the PBX to each individual extension and device, before this can operate as a usable system. See Grandstream's documentation for more info.
Website: UCM 6200 Series
Contents |
Login into your device
- Connect a computer to the same network as the UCM6202
- Ensure the UCM 6202 is properly powered on and displays the IP address on the LCD screen
- Open a web browser on the computer and enter the displayed IP address into the search bar in the following format http(s)://ipaddress:portnumber. The default protocol is HTTPS and the default port number is 8089.
- The Web portal should be shown (see figure below). The default username is "admin". The default password (P/W) is a random string indicated on a sticker on the back of the unit for Grandstream hardware manufactured after 2017; earlier hardware revisions had a default password of "admin".
Creating a trunk
To create a new VoIP.ms trunk head to Extension/Trunk>> VoIP trunks, from the left panel. In this section, you can choose to create a SIP or an IAX trunk.
SIP Trunk
To create a SIP trunk you only need to fill some basic information.
- Type: Register SIP trunk
- Provider Name: VoIP.ms (any name can be used)
- Host Name: Type any of our servers, i.e. toronto5.voip.ms
- Keep trunk CID: Enable it, if you want the trunk to send its own CID number or disable it if your extensions are going to send their own CID number
- Username: 100000 (replace with your main VoIP.ms account number or sub-account name)
- Password: ********* (replace with your main SIP/IAX password or sub-account password)
Note: Bear in mind to use the same VoIP server your VoIP number is using. You can check what VoIP server is your VoIP number using, from your VoIP.ms customer portal at DID Numbers>> Manage DIDs, under POP column. You can choose any server you want, as long as the one in your portal and the one in this field matches, otherwise, incoming calls won't ring.
Click Save button, do not click Apply Changes yet.
Extra SIP settings
Once your basic SIP trunk has been created we will proceed to improve some settings on it, click on the edit icon for your trunk.
SIP Headers
To send the SIP "FROM" header as we require, make:
"fromuser:" 100000 (replace with your main VoIP.ms account number or sub-account name)
Audio codecs
Click at "Advanced Settings" and at "Codec Preference" use only the supported codecs by VoIP.ms: G.729, PCMU & GSM, in this order.
Trunk Caller ID
You can use PAI (P-Asserted-Identity) header if you want to send the CID name & number from your trunk, you only need to use this format:
"CallerIDName"<CallerIDNumber>
Note: Enabling the PAI header but leaving the field blank (null) will force the pbx to use the CID priority rules from Grandstream thus enabling you to passthorugh the CID from the extensions
Note: No Caller ID Number must be set at your VoIP.ms portal or it will override the one sent by your trunk
NAT Keep Alive
In order to avoid your modem closing your local SIP ports, enable:
*Enable Heartbeat Detection: Enabled
*Heartbeat Frequency: 50
Finally, click Save button and at this stage, you can also click on Apply Changes. Your trunk should be shown as Registered from your VoIP.ms dashboard, however, no calls will work until you set up your outbound and inbound routes.
Call Encryption TLS/SRTP
In order to use TLS along with your UCM please follow these steps:
1. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled.
Bear in mind, if this setting is enabled and your device sends UDP/TCP, RTP you will be rejected with error code 488.
Enable this setting for the Main Account at Main Menu>> Account settings>> Advanced tab.
For a sub-account enable it at Sub accounts>> Manage sub-accounts by clicking on the orange icon with a pen and finally click at "Advanced Options (Click here to display)".
2. Now that your account/sub-account has this setting enabled, your device only needs to send TLS and SRTP.
Go to Extension/Trunk>> VoIP Trunks and click on "Edit trunk". In this section make sure you have the following settings:
*Host Name: toronto5.voip.ms:5061 (Use the same server your phone number is at, you can check it out from your customer portal at Manage DIDs section).
*Transport: TLS
Note: When using TLS is very important to specify the number of the server, in case the server's name doesn't have the number "1" included, you need to add it. Adding any of the SIP ports 5061/5081/42873 at the end of the Hostname is also required.
Go to Advanced Settings and set SRTP to "Enabled and forced"
IAX2 Trunk
To create an IAX2 trunk just fill the following information:
- Type: Register IAX trunk
- Provider Name: VoIP.ms (any name can be used)
- Host Name: Type any of our servers, i.e. toronto5.voip.ms
- Keep trunk CID: Enable it, if you want the trunk to send its own CID number or disable it if your extensions are going to send their own CID number
- Caller ID: Type the CID number your trunk will be sending.
- Caller ID Name: Type the name your trunk will be sending.
- Username: 100000 (replace with your main VoIP.ms account number or sub-account name)
- Password: ********* (replace with your main SIP/IAX password or sub-account password)
Note: Bear in mind to use the same VoIP server your VoIP number is using. You can check what VoIP server is your VoIP number using, from your VoIP.ms customer portal at DID Numbers>> Manage DIDs, under POP column. You can choose any server you want, as long as the one in your portal and the one in this field matches, otherwise, incoming calls won't ring.
Click Save button, do not click Apply Changes yet.
Extra IAX settings
Once your basic IAX trunk has been created we will proceed to improve some settings on it, click on the edit icon for your trunk.
Audio codecs
Click at "Advanced Settings" and at "Codec Preference" use only the supported codecs by VoIP.ms: G.729, PCMU & GSM, in this order.
NAT Keep Alive
In order to avoid your modem closing your local ports, enable:
*Enable Heartbeat Detection: Enabled
*Heartbeat Frequency: 50
Finally, click Save button and at this stage, you can also click on Apply Changes. Your trunk should be shown as Registered from your VoIP.ms dashboard, however, no calls will work until you set up your outbound and inbound routes.
Creating your outbound route
Outbound routes are the ones in charge of making match your dialing pattern and send your call through the proper trunk
VoIP.ms suggest to include the following patterns into your outbound route:
_1NXXXXXXXXX _NXXXXXXXXX _4XXX _00. _011. _033. _044.
All your different dial patterns must be prefixed by the character "_"
To create your outbound routes click on Extension/Trunk>> Outbound routes, from the left panel and click on "Add"
In this section you only need to fill the following fields:
- Calling Rule Name: Any name you want for this route.
- Pattern: The desired patter your callers need to dial. You can use the suggested dial pattern by VoIP.ms shown above.
- Trunk: Choose your VoIP.ms trunk on where your call will be sent through.
- Privilege Level: Choose the desired privilege your extensions must have to be able to use this route. If you use our dial pattern we suggest to set it to "International" since you will be able to place international calls. Bear in mind that an extension with an inferior privilege won't be able to use this route. You can hover over the field's name to gather more information.
Note: If you want to include a dial-out prefix, you can type it after the "_" character in your dial patterns. This number will need to be stripped off, you can do this by using the "Strip" field, you can choose how many digits you can strip after the "_" character.
For example: If you want to use "9" to dial out, then your pattern will need to be _9NXXXXXXXXX To strip off this number "9" when dialing out, you will need to set the "Strip" field to "1", this way only one character (in this case number "9") will be stripped off.
Creating your inbound route
Thanks to your inbound routes you can use only one single trunk to receive all the incoming calls from all your phone numbers. This way you don't need to use more than one trunk for your phone numbers, inbound routes will receive all of them and route them into the proper destination in your UCM.
Note: We do not suggest using more than one VoIP.ms trunk on the same device.
From the left panel, head to Extension/Trunk>> Inbound Routes and click on "Add".
In this section, fill the following fields:
- Trunks: Choose your VoIP.ms trunk, on where the incoming calls should be sent by us.
- Pattern: Type your VoIP.ms DID phone number exactly as it is shown under "Manage DIDs" section from your customer portal. This number should be prefixed by the character "_". Please avoid using Wildcards and ensure to use full numbers, otherwise the inbound pattern will have conflicts.
- Default Destination: Choose the default destination on where your PBX should send your incoming calls to. This could be an internal extension, an IVR, a ring group and so on; managed locally by your PBX.
Note: Remember to send your phone number in the "TO" SIP header, this way your PBx will match it with your inbound routes. You can do this very easily by setting Device Type setting from "ATA adapter, IP phone or Softphone" to "IP PBX Server, Asterisk or Softswitch".
You will find this setting from your VoIP.ms customer portal at Main Menu>> Account settings>> Inbound settings, if you're using the main account or at Sub accounts>> Manage Sub accounts and by clicking on the orange icon with a pen, if you're using a sub-account
- Using the main account
- Using a subaccount
External links
Guides and manuals
As a PBX, the Grandstream UCM6200 series is a powerful device with a long list of options; the manufacturer's full administrators guide runs more than four hundred pages:
